mirror of
https://github.com/holub/mame
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1127 lines
34 KiB
C++
1127 lines
34 KiB
C++
// license:BSD-3-Clause
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// copyright-holders:Aaron Giles
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/***************************************************************************
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sound.c
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Core sound functions and definitions.
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***************************************************************************/
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#include "emu.h"
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#include "speaker.h"
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#include "emuopts.h"
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#include "osdepend.h"
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#include "config.h"
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#include "wavwrite.h"
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//**************************************************************************
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// DEBUGGING
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//**************************************************************************
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#define VERBOSE (0)
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#define VPRINTF(x) do { if (VERBOSE) osd_printf_debug x; } while (0)
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//**************************************************************************
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// CONSTANTS
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//**************************************************************************
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//**************************************************************************
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// GLOBAL VARIABLES
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//**************************************************************************
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const attotime sound_manager::STREAMS_UPDATE_ATTOTIME = attotime::from_hz(STREAMS_UPDATE_FREQUENCY);
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//**************************************************************************
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// INITIALIZATION
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//**************************************************************************
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//-------------------------------------------------
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// sound_stream - constructor
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//-------------------------------------------------
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sound_stream::sound_stream(device_t &device, int inputs, int outputs, int sample_rate, stream_update_delegate callback)
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: m_device(device),
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m_next(nullptr),
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m_sample_rate(sample_rate),
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m_new_sample_rate(0),
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m_attoseconds_per_sample(0),
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m_max_samples_per_update(0),
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m_input(inputs),
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m_input_array(inputs),
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m_resample_bufalloc(0),
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m_output(outputs),
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m_output_array(outputs),
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m_output_bufalloc(0),
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m_output_sampindex(0),
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m_output_update_sampindex(0),
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m_output_base_sampindex(0),
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m_callback(std::move(callback))
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{
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// get the device's sound interface
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device_sound_interface *sound;
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if (!device.interface(sound))
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throw emu_fatalerror("Attempted to create a sound_stream with a non-sound device");
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if(m_callback.isnull())
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m_callback = stream_update_delegate(&device_sound_interface::sound_stream_update,(device_sound_interface *)sound);
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// create a unique tag for saving
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std::string state_tag = string_format("%d", m_device.machine().sound().m_stream_list.size());
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m_device.machine().save().save_item(&m_device, "stream", state_tag.c_str(), 0, NAME(m_sample_rate));
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m_device.machine().save().register_postload(save_prepost_delegate(FUNC(sound_stream::postload), this));
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// save the gain of each input and output
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for (unsigned int inputnum = 0; inputnum < m_input.size(); inputnum++)
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{
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m_device.machine().save().save_item(&m_device, "stream", state_tag.c_str(), inputnum, NAME(m_input[inputnum].m_gain));
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m_device.machine().save().save_item(&m_device, "stream", state_tag.c_str(), inputnum, NAME(m_input[inputnum].m_user_gain));
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}
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for (unsigned int outputnum = 0; outputnum < m_output.size(); outputnum++)
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{
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m_output[outputnum].m_stream = this;
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m_device.machine().save().save_item(&m_device, "stream", state_tag.c_str(), outputnum, NAME(m_output[outputnum].m_gain));
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}
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// Mark synchronous streams as such
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m_synchronous = m_sample_rate == STREAM_SYNC;
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if (m_synchronous)
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{
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m_sample_rate = 0;
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m_sync_timer = m_device.machine().scheduler().timer_alloc(timer_expired_delegate(FUNC(sound_stream::sync_update), this));
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}
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else
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m_sync_timer = nullptr;
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// force an update to the sample rates; this will cause everything to be recomputed
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// and will generate the initial resample buffers for our inputs
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recompute_sample_rate_data();
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// set up the initial output buffer positions now that we have data
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m_output_base_sampindex = -m_max_samples_per_update;
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}
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//-------------------------------------------------
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// sample_time - return the emulation time of the
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// next sample to be generated on the stream
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//-------------------------------------------------
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attotime sound_stream::sample_time() const
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{
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return attotime(m_device.machine().sound().last_update().seconds(), 0) + attotime(0, m_output_sampindex * m_attoseconds_per_sample);
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}
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//-------------------------------------------------
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// user_gain - return the user-controllable gain
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// on a given stream's input
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//-------------------------------------------------
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float sound_stream::user_gain(int inputnum) const
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{
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assert(inputnum >= 0 && inputnum < m_input.size());
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return float(m_input[inputnum].m_user_gain) / 256.0f;
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}
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//-------------------------------------------------
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// input_gain - return the input gain on a
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// given stream's input
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//-------------------------------------------------
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float sound_stream::input_gain(int inputnum) const
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{
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assert(inputnum >= 0 && inputnum < m_input.size());
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return float(m_input[inputnum].m_gain) / 256.0f;
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}
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//-------------------------------------------------
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// output_gain - return the output gain on a
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// given stream's output
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//-------------------------------------------------
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float sound_stream::output_gain(int outputnum) const
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{
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assert(outputnum >= 0 && outputnum < m_output.size());
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return float(m_output[outputnum].m_gain) / 256.0f;
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}
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//-------------------------------------------------
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// input_name - return the original input gain
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// on a given stream's input
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//-------------------------------------------------
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std::string sound_stream::input_name(int inputnum) const
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{
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std::ostringstream str;
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// start with our device name and tag
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assert(inputnum >= 0 && inputnum < m_input.size());
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util::stream_format(str, "%s '%s': ", m_device.name(), m_device.tag());
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// if we have a source, indicate where the sound comes from by device name and tag
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if (m_input[inputnum].m_source != nullptr && m_input[inputnum].m_source->m_stream != nullptr)
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{
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device_t &source = m_input[inputnum].m_source->m_stream->device();
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util::stream_format(str, "%s '%s'", source.name(), source.tag());
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// get the sound interface; if there is more than 1 output we need to figure out which one
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device_sound_interface *sound;
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if (source.interface(sound) && sound->outputs() > 1)
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{
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// iterate over outputs until we find the stream that matches our source
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// then look for a match on the output number
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sound_stream *outstream;
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int streamoutputnum;
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for (int outputnum = 0; (outstream = sound->output_to_stream_output(outputnum, streamoutputnum)) != nullptr; outputnum++)
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if (outstream == m_input[inputnum].m_source->m_stream && m_input[inputnum].m_source == &outstream->m_output[streamoutputnum])
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{
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util::stream_format(str, " Ch.%d", outputnum);
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break;
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}
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}
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}
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return str.str();
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}
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//-------------------------------------------------
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// input_source_device - return the device
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// attached as a given input's source
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//-------------------------------------------------
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device_t *sound_stream::input_source_device(int inputnum) const
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{
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assert(inputnum >= 0 && inputnum < m_input.size());
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return (m_input[inputnum].m_source != nullptr) ? &m_input[inputnum].m_source->m_stream->device() : nullptr;
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}
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//-------------------------------------------------
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// input_source_device - return the output number
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// attached as a given input's source
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//-------------------------------------------------
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int sound_stream::input_source_outputnum(int inputnum) const
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{
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assert(inputnum >= 0 && inputnum < m_input.size());
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return (m_input[inputnum].m_source != nullptr) ? (m_input[inputnum].m_source - &m_input[inputnum].m_source->m_stream->m_output[0]) : -1;
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}
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//-------------------------------------------------
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// set_input - configure a stream's input
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//-------------------------------------------------
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void sound_stream::set_input(int index, sound_stream *input_stream, int output_index, float gain)
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{
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VPRINTF(("stream_set_input(%p, '%s', %d, %p, %d, %f)\n", (void *)this, m_device.tag(),
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index, (void *)input_stream, output_index, (double) gain));
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// make sure it's a valid input
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if (index >= m_input.size())
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fatalerror("stream_set_input attempted to configure non-existant input %d (%d max)\n", index, int(m_input.size()));
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// make sure it's a valid output
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if (input_stream != nullptr && output_index >= input_stream->m_output.size())
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fatalerror("stream_set_input attempted to use a non-existant output %d (%d max)\n", output_index, int(m_output.size()));
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// if this input is already wired, update the dependent info
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stream_input &input = m_input[index];
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if (input.m_source != nullptr)
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input.m_source->m_dependents--;
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// wire it up
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input.m_source = (input_stream != nullptr) ? &input_stream->m_output[output_index] : nullptr;
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input.m_gain = int(0x100 * gain);
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input.m_user_gain = 0x100;
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// update the dependent info
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if (input.m_source != nullptr)
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input.m_source->m_dependents++;
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// update sample rates now that we know the input
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recompute_sample_rate_data();
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}
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//-------------------------------------------------
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// update - force a stream to update to
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// the current emulated time
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//-------------------------------------------------
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void sound_stream::update()
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{
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// determine the number of samples since the start of this second
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attotime time = m_device.machine().time();
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s32 update_sampindex = s32(time.attoseconds() / m_attoseconds_per_sample);
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// if we're ahead of the last update, then adjust upwards
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attotime last_update = m_device.machine().sound().last_update();
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if (time.seconds() > last_update.seconds())
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{
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assert(time.seconds() == last_update.seconds() + 1);
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update_sampindex += m_sample_rate;
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}
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// if we're behind the last update, then adjust downwards
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if (time.seconds() < last_update.seconds())
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{
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assert(time.seconds() == last_update.seconds() - 1);
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update_sampindex -= m_sample_rate;
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}
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if (update_sampindex <= m_output_sampindex)
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return;
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// generate samples to get us up to the appropriate time
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g_profiler.start(PROFILER_SOUND);
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assert(m_output_sampindex - m_output_base_sampindex >= 0);
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assert(update_sampindex - m_output_base_sampindex <= m_output_bufalloc);
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generate_samples(update_sampindex - m_output_sampindex);
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g_profiler.stop();
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// remember this info for next time
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m_output_sampindex = update_sampindex;
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}
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void sound_stream::sync_update(void *, s32)
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{
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update();
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attotime time = m_device.machine().time();
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attoseconds_t next_edge = m_attoseconds_per_sample - (time.attoseconds() % m_attoseconds_per_sample);
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m_sync_timer->adjust(attotime(0, next_edge));
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}
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//-------------------------------------------------
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// output_since_last_update - return a pointer to
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// the output buffer and the number of samples
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// since the last global update
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//-------------------------------------------------
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const stream_sample_t *sound_stream::output_since_last_update(int outputnum, int &numsamples)
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{
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// force an update on the stream
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update();
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// compute the number of samples and a pointer to the output buffer
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numsamples = m_output_sampindex - m_output_update_sampindex;
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return &m_output[outputnum].m_buffer[m_output_update_sampindex - m_output_base_sampindex];
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}
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//-------------------------------------------------
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// set_sample_rate - set the sample rate on a
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// given stream
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//-------------------------------------------------
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void sound_stream::set_sample_rate(int new_rate)
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{
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// we will update this on the next global update
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if (new_rate != sample_rate())
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m_new_sample_rate = new_rate;
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}
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//-------------------------------------------------
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// set_user_gain - set the user-controllable gain
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// on a given stream's input
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//-------------------------------------------------
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void sound_stream::set_user_gain(int inputnum, float gain)
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{
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update();
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assert(inputnum >= 0 && inputnum < m_input.size());
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m_input[inputnum].m_user_gain = int(0x100 * gain);
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}
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//-------------------------------------------------
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// set_input_gain - set the input gain on a
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// given stream's input
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//-------------------------------------------------
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void sound_stream::set_input_gain(int inputnum, float gain)
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{
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update();
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assert(inputnum >= 0 && inputnum < m_input.size());
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m_input[inputnum].m_gain = int(0x100 * gain);
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}
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//-------------------------------------------------
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// set_output_gain - set the output gain on a
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// given stream's output
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//-------------------------------------------------
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void sound_stream::set_output_gain(int outputnum, float gain)
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{
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update();
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assert(outputnum >= 0 && outputnum < m_output.size());
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m_output[outputnum].m_gain = int(0x100 * gain);
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}
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//-------------------------------------------------
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// update_with_accounting - do a regular update,
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// but also do periodic accounting
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//-------------------------------------------------
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void sound_stream::update_with_accounting(bool second_tick)
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{
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// do the normal update
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update();
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// if we've ticked over another second, adjust all the counters that are relative to
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// the current second
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s32 output_bufindex = m_output_sampindex - m_output_base_sampindex;
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if (second_tick)
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{
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m_output_sampindex -= m_sample_rate;
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m_output_base_sampindex -= m_sample_rate;
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}
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// note our current output sample
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m_output_update_sampindex = m_output_sampindex;
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// if we don't have enough output buffer space to hold two updates' worth of samples,
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// we need to shuffle things down
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if (m_output_bufalloc - output_bufindex < 2 * m_max_samples_per_update)
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{
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s32 samples_to_lose = output_bufindex - m_max_samples_per_update;
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if (samples_to_lose > 0)
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{
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// if we have samples to move, do so for each output
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if (output_bufindex > 0)
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for (auto & output : m_output)
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{
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memmove(&output.m_buffer[0], &output.m_buffer[samples_to_lose], sizeof(output.m_buffer[0]) * (output_bufindex - samples_to_lose));
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}
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// update the base position
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m_output_base_sampindex += samples_to_lose;
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}
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}
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}
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//-------------------------------------------------
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// apply_sample_rate_changes - if there is a
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// pending sample rate change, apply it now
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//-------------------------------------------------
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void sound_stream::apply_sample_rate_changes()
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{
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// skip if nothing to do
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if (m_new_sample_rate == 0)
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return;
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// update to the new rate and remember the old rate
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u32 old_rate = m_sample_rate;
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m_sample_rate = m_new_sample_rate;
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m_new_sample_rate = 0;
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// recompute all the data
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recompute_sample_rate_data();
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// reset our sample indexes to the current time
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m_output_sampindex = s64(m_output_sampindex) * s64(m_sample_rate) / old_rate;
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m_output_update_sampindex = s64(m_output_update_sampindex) * s64(m_sample_rate) / old_rate;
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m_output_base_sampindex = m_output_sampindex - m_max_samples_per_update;
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// clear out the buffer
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for (auto & elem : m_output)
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memset(&elem.m_buffer[0], 0, m_max_samples_per_update * sizeof(elem.m_buffer[0]));
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}
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//-------------------------------------------------
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// recompute_sample_rate_data - recompute sample
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// rate data, and all streams that are affected
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// by this stream
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//-------------------------------------------------
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void sound_stream::recompute_sample_rate_data()
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{
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if (m_synchronous)
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{
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m_sample_rate = 0;
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// When synchronous, pick the sample rate for the inputs, if any
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for (auto & input : m_input)
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{
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if (input.m_source != nullptr)
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{
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if (!m_sample_rate)
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m_sample_rate = input.m_source->m_stream->m_sample_rate;
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else if (m_sample_rate != input.m_source->m_stream->m_sample_rate)
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throw emu_fatalerror("Incompatible sample rates as input of a synchronous stream: %d and %d\n", m_sample_rate, input.m_source->m_stream->m_sample_rate);
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}
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}
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if (!m_sample_rate)
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m_sample_rate = 1000;
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}
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// recompute the timing parameters
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attoseconds_t update_attoseconds = m_device.machine().sound().update_attoseconds();
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m_attoseconds_per_sample = ATTOSECONDS_PER_SECOND / m_sample_rate;
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m_max_samples_per_update = (update_attoseconds + m_attoseconds_per_sample - 1) / m_attoseconds_per_sample;
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// update resample and output buffer sizes
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allocate_resample_buffers();
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allocate_output_buffers();
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// iterate over each input
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for (auto & input : m_input)
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{
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// if we have a source, see if its sample rate changed
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if (input.m_source != nullptr)
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{
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// okay, we have a new sample rate; recompute the latency to be the maximum
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// sample period between us and our input
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attoseconds_t new_attosecs_per_sample = ATTOSECONDS_PER_SECOND / input.m_source->m_stream->m_sample_rate;
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attoseconds_t latency = std::max(new_attosecs_per_sample, m_attoseconds_per_sample);
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// if the input stream's sample rate is lower, we will use linear interpolation
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// this requires an extra sample from the source
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if (input.m_source->m_stream->m_sample_rate < m_sample_rate)
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latency += new_attosecs_per_sample;
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// if our sample rates match exactly, we don't need any latency
|
|
else if (input.m_source->m_stream->m_sample_rate == m_sample_rate)
|
|
latency = 0;
|
|
|
|
// we generally don't want to tweak the latency, so we just keep the greatest
|
|
// one we've computed thus far
|
|
input.m_latency_attoseconds = std::max(input.m_latency_attoseconds, latency);
|
|
assert(input.m_latency_attoseconds < update_attoseconds);
|
|
}
|
|
}
|
|
|
|
// If synchronous, prime the timer
|
|
if (m_synchronous)
|
|
{
|
|
attotime time = m_device.machine().time();
|
|
attoseconds_t next_edge = m_attoseconds_per_sample - (time.attoseconds() % m_attoseconds_per_sample);
|
|
m_sync_timer->adjust(attotime(0, next_edge));
|
|
}
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// allocate_resample_buffers - recompute the
|
|
// resample buffer sizes and expand if necessary
|
|
//-------------------------------------------------
|
|
|
|
void sound_stream::allocate_resample_buffers()
|
|
{
|
|
// compute the target number of samples
|
|
s32 bufsize = 2 * m_max_samples_per_update;
|
|
|
|
// if we don't have enough room, allocate more
|
|
if (m_resample_bufalloc < bufsize)
|
|
{
|
|
// this becomes the new allocation size
|
|
m_resample_bufalloc = bufsize;
|
|
|
|
// iterate over outputs and realloc their buffers
|
|
for (auto & elem : m_input) {
|
|
unsigned int old_size = elem.m_resample.size();
|
|
elem.m_resample.resize(m_resample_bufalloc);
|
|
memset(&elem.m_resample[old_size], 0, (m_resample_bufalloc - old_size)*sizeof(elem.m_resample[0]));
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// allocate_output_buffers - recompute the
|
|
// output buffer sizes and expand if necessary
|
|
//-------------------------------------------------
|
|
|
|
void sound_stream::allocate_output_buffers()
|
|
{
|
|
// if we don't have enough room, allocate more
|
|
s32 bufsize = OUTPUT_BUFFER_UPDATES * m_max_samples_per_update;
|
|
if (m_output_bufalloc < bufsize)
|
|
{
|
|
// this becomes the new allocation size
|
|
m_output_bufalloc = bufsize;
|
|
|
|
// iterate over outputs and realloc their buffers
|
|
for (auto & elem : m_output) {
|
|
unsigned int old_size = elem.m_buffer.size();
|
|
elem.m_buffer.resize(m_output_bufalloc);
|
|
memset(&elem.m_buffer[old_size], 0, (m_output_bufalloc - old_size)*sizeof(elem.m_buffer[0]));
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// postload - save/restore callback
|
|
//-------------------------------------------------
|
|
|
|
void sound_stream::postload()
|
|
{
|
|
// recompute the same rate information
|
|
recompute_sample_rate_data();
|
|
|
|
// make sure our output buffers are fully cleared
|
|
for (auto & elem : m_output)
|
|
memset(&elem.m_buffer[0], 0, m_output_bufalloc * sizeof(elem.m_buffer[0]));
|
|
|
|
// recompute the sample indexes to make sense
|
|
m_output_sampindex = m_device.machine().sound().last_update().attoseconds() / m_attoseconds_per_sample;
|
|
m_output_update_sampindex = m_output_sampindex;
|
|
m_output_base_sampindex = m_output_sampindex - m_max_samples_per_update;
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// generate_samples - generate the requested
|
|
// number of samples for a stream, making sure
|
|
// all inputs have the appropriate number of
|
|
// samples generated
|
|
//-------------------------------------------------
|
|
|
|
void sound_stream::generate_samples(int samples)
|
|
{
|
|
stream_sample_t **inputs = nullptr;
|
|
stream_sample_t **outputs = nullptr;
|
|
|
|
VPRINTF(("generate_samples(%p, %d)\n", (void *) this, samples));
|
|
assert(samples > 0);
|
|
|
|
// ensure all inputs are up to date and generate resampled data
|
|
for (unsigned int inputnum = 0; inputnum < m_input.size(); inputnum++)
|
|
{
|
|
// update the stream to the current time
|
|
stream_input &input = m_input[inputnum];
|
|
if (input.m_source != nullptr)
|
|
input.m_source->m_stream->update();
|
|
|
|
// generate the resampled data
|
|
m_input_array[inputnum] = generate_resampled_data(input, samples);
|
|
}
|
|
|
|
if (!m_input.empty())
|
|
{
|
|
inputs = &m_input_array[0];
|
|
}
|
|
|
|
// loop over all outputs and compute the output pointer
|
|
for (unsigned int outputnum = 0; outputnum < m_output.size(); outputnum++)
|
|
{
|
|
stream_output &output = m_output[outputnum];
|
|
m_output_array[outputnum] = &output.m_buffer[m_output_sampindex - m_output_base_sampindex];
|
|
}
|
|
|
|
if (!m_output.empty())
|
|
{
|
|
outputs = &m_output_array[0];
|
|
}
|
|
|
|
// run the callback
|
|
VPRINTF((" callback(%p, %d)\n", (void *)this, samples));
|
|
m_callback(*this, inputs, outputs, samples);
|
|
VPRINTF((" callback done\n"));
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// generate_resampled_data - generate the
|
|
// resample buffer for a given input
|
|
//-------------------------------------------------
|
|
|
|
stream_sample_t *sound_stream::generate_resampled_data(stream_input &input, u32 numsamples)
|
|
{
|
|
// if we don't have an output to pull data from, generate silence
|
|
stream_sample_t *dest = &input.m_resample[0];
|
|
if (input.m_source == nullptr)
|
|
{
|
|
memset(dest, 0, numsamples * sizeof(*dest));
|
|
return &input.m_resample[0];
|
|
}
|
|
|
|
// grab data from the output
|
|
stream_output &output = *input.m_source;
|
|
sound_stream &input_stream = *output.m_stream;
|
|
s64 gain = (input.m_gain * input.m_user_gain * output.m_gain) >> 16;
|
|
|
|
// determine the time at which the current sample begins, accounting for the
|
|
// latency we calculated between the input and output streams
|
|
attoseconds_t basetime = m_output_sampindex * m_attoseconds_per_sample - input.m_latency_attoseconds;
|
|
|
|
// now convert that time into a sample in the input stream
|
|
s32 basesample;
|
|
if (basetime >= 0)
|
|
basesample = basetime / input_stream.m_attoseconds_per_sample;
|
|
else
|
|
basesample = -(-basetime / input_stream.m_attoseconds_per_sample) - 1;
|
|
|
|
// compute a source pointer to the first sample
|
|
assert(basesample >= input_stream.m_output_base_sampindex);
|
|
stream_sample_t *source = &output.m_buffer[basesample - input_stream.m_output_base_sampindex];
|
|
|
|
// determine the current fraction of a sample, expressed as a fraction of FRAC_ONE
|
|
// (Note: this formula is valid as long as input_stream.m_attoseconds_per_sample significantly exceeds FRAC_ONE > attoseconds = 4.2E-12 s)
|
|
u32 basefrac = (basetime - basesample * input_stream.m_attoseconds_per_sample) / ((input_stream.m_attoseconds_per_sample + FRAC_ONE - 1) >> FRAC_BITS);
|
|
assert(basefrac < FRAC_ONE);
|
|
|
|
// compute the stepping fraction
|
|
u32 step = (u64(input_stream.m_sample_rate) << FRAC_BITS) / m_sample_rate;
|
|
|
|
// if we have equal sample rates, we just need to copy
|
|
if (step == FRAC_ONE)
|
|
{
|
|
while (numsamples--)
|
|
{
|
|
// compute the sample
|
|
s64 sample = *source++;
|
|
*dest++ = (sample * gain) >> 8;
|
|
}
|
|
}
|
|
|
|
// input is undersampled: point sample except where our sample period covers a boundary
|
|
else if (step < FRAC_ONE)
|
|
{
|
|
while (numsamples != 0)
|
|
{
|
|
// fill in with point samples until we hit a boundary
|
|
int nextfrac;
|
|
while ((nextfrac = basefrac + step) < FRAC_ONE && numsamples--)
|
|
{
|
|
*dest++ = (source[0] * gain) >> 8;
|
|
basefrac = nextfrac;
|
|
}
|
|
|
|
// if we're done, we're done
|
|
if (s32(numsamples--) < 0)
|
|
break;
|
|
|
|
// compute starting and ending fractional positions
|
|
int startfrac = basefrac >> (FRAC_BITS - 12);
|
|
int endfrac = nextfrac >> (FRAC_BITS - 12);
|
|
|
|
// blend between the two samples accordingly
|
|
s64 sample = (s64(source[0]) * (0x1000 - startfrac) + s64(source[1]) * (endfrac - 0x1000)) / (endfrac - startfrac);
|
|
*dest++ = (sample * gain) >> 8;
|
|
|
|
// advance
|
|
basefrac = nextfrac & FRAC_MASK;
|
|
source++;
|
|
}
|
|
}
|
|
|
|
// input is oversampled: sum the energy
|
|
else
|
|
{
|
|
// use 8 bits to allow some extra headroom
|
|
int smallstep = step >> (FRAC_BITS - 8);
|
|
while (numsamples--)
|
|
{
|
|
s64 remainder = smallstep;
|
|
int tpos = 0;
|
|
|
|
// compute the sample
|
|
s64 scale = (FRAC_ONE - basefrac) >> (FRAC_BITS - 8);
|
|
s64 sample = s64(source[tpos++]) * scale;
|
|
remainder -= scale;
|
|
while (remainder > 0x100)
|
|
{
|
|
sample += s64(source[tpos++]) * s64(0x100);
|
|
remainder -= 0x100;
|
|
}
|
|
sample += s64(source[tpos]) * remainder;
|
|
sample /= smallstep;
|
|
|
|
*dest++ = (sample * gain) >> 8;
|
|
|
|
// advance
|
|
basefrac += step;
|
|
source += basefrac >> FRAC_BITS;
|
|
basefrac &= FRAC_MASK;
|
|
}
|
|
}
|
|
|
|
return &input.m_resample[0];
|
|
}
|
|
|
|
|
|
|
|
//**************************************************************************
|
|
// STREAM INPUT
|
|
//**************************************************************************
|
|
|
|
//-------------------------------------------------
|
|
// stream_input - constructor
|
|
//-------------------------------------------------
|
|
|
|
sound_stream::stream_input::stream_input()
|
|
: m_source(nullptr),
|
|
m_latency_attoseconds(0),
|
|
m_gain(0x100),
|
|
m_user_gain(0x100)
|
|
{
|
|
}
|
|
|
|
|
|
|
|
//**************************************************************************
|
|
// STREAM OUTPUT
|
|
//**************************************************************************
|
|
|
|
//-------------------------------------------------
|
|
// stream_output - constructor
|
|
//-------------------------------------------------
|
|
|
|
sound_stream::stream_output::stream_output()
|
|
: m_stream(nullptr),
|
|
m_dependents(0),
|
|
m_gain(0x100)
|
|
{
|
|
}
|
|
|
|
|
|
|
|
//**************************************************************************
|
|
// SOUND MANAGER
|
|
//**************************************************************************
|
|
|
|
//-------------------------------------------------
|
|
// sound_manager - constructor
|
|
//-------------------------------------------------
|
|
|
|
sound_manager::sound_manager(running_machine &machine)
|
|
: m_machine(machine),
|
|
m_update_timer(nullptr),
|
|
m_finalmix_leftover(0),
|
|
m_finalmix(machine.sample_rate()),
|
|
m_leftmix(machine.sample_rate()),
|
|
m_rightmix(machine.sample_rate()),
|
|
m_muted(0),
|
|
m_attenuation(0),
|
|
m_nosound_mode(machine.osd().no_sound()),
|
|
m_wavfile(nullptr),
|
|
m_update_attoseconds(STREAMS_UPDATE_ATTOTIME.attoseconds()),
|
|
m_last_update(attotime::zero)
|
|
{
|
|
// get filename for WAV file or AVI file if specified
|
|
const char *wavfile = machine.options().wav_write();
|
|
const char *avifile = machine.options().avi_write();
|
|
|
|
// handle -nosound and lower sample rate if not recording WAV or AVI
|
|
if (m_nosound_mode && wavfile[0] == 0 && avifile[0] == 0)
|
|
machine.m_sample_rate = 11025;
|
|
|
|
// count the mixers
|
|
#if VERBOSE
|
|
mixer_interface_iterator iter(machine.root_device());
|
|
VPRINTF(("total mixers = %d\n", iter.count()));
|
|
#endif
|
|
|
|
// register callbacks
|
|
machine.configuration().config_register("mixer", config_load_delegate(&sound_manager::config_load, this), config_save_delegate(&sound_manager::config_save, this));
|
|
machine.add_notifier(MACHINE_NOTIFY_PAUSE, machine_notify_delegate(&sound_manager::pause, this));
|
|
machine.add_notifier(MACHINE_NOTIFY_RESUME, machine_notify_delegate(&sound_manager::resume, this));
|
|
machine.add_notifier(MACHINE_NOTIFY_RESET, machine_notify_delegate(&sound_manager::reset, this));
|
|
machine.add_notifier(MACHINE_NOTIFY_EXIT, machine_notify_delegate(&sound_manager::stop_recording, this));
|
|
|
|
// register global states
|
|
machine.save().save_item(NAME(m_last_update));
|
|
|
|
// set the starting attenuation
|
|
set_attenuation(machine.options().volume());
|
|
|
|
// start the periodic update flushing timer
|
|
m_update_timer = machine.scheduler().timer_alloc(timer_expired_delegate(FUNC(sound_manager::update), this));
|
|
m_update_timer->adjust(STREAMS_UPDATE_ATTOTIME, 0, STREAMS_UPDATE_ATTOTIME);
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// sound_manager - destructor
|
|
//-------------------------------------------------
|
|
|
|
sound_manager::~sound_manager()
|
|
{
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// start_recording - begin audio recording
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::start_recording()
|
|
{
|
|
// open the output WAV file if specified
|
|
const char *wavfile = machine().options().wav_write();
|
|
if (wavfile[0] != 0 && m_wavfile == nullptr)
|
|
m_wavfile = wav_open(wavfile, machine().sample_rate(), 2);
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// stop_recording - end audio recording
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::stop_recording()
|
|
{
|
|
// close any open WAV file
|
|
if (m_wavfile != nullptr)
|
|
wav_close(m_wavfile);
|
|
m_wavfile = nullptr;
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// stream_alloc - allocate a new stream
|
|
//-------------------------------------------------
|
|
|
|
sound_stream *sound_manager::stream_alloc(device_t &device, int inputs, int outputs, int sample_rate, stream_update_delegate callback)
|
|
{
|
|
m_stream_list.push_back(std::make_unique<sound_stream>(device, inputs, outputs, sample_rate, callback));
|
|
return m_stream_list.back().get();
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// set_attenuation - set the global volume
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::set_attenuation(int attenuation)
|
|
{
|
|
m_attenuation = attenuation;
|
|
machine().osd().set_mastervolume(m_muted ? -32 : m_attenuation);
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// indexed_mixer_input - return the mixer
|
|
// device and input index of the global mixer
|
|
// input
|
|
//-------------------------------------------------
|
|
|
|
bool sound_manager::indexed_mixer_input(int index, mixer_input &info) const
|
|
{
|
|
// scan through the mixers until we find the indexed input
|
|
for (device_mixer_interface &mixer : mixer_interface_iterator(machine().root_device()))
|
|
{
|
|
if (index < mixer.inputs())
|
|
{
|
|
info.mixer = &mixer;
|
|
info.stream = mixer.input_to_stream_input(index, info.inputnum);
|
|
assert(info.stream != nullptr);
|
|
return true;
|
|
}
|
|
index -= mixer.inputs();
|
|
}
|
|
|
|
// didn't locate
|
|
info.mixer = nullptr;
|
|
return false;
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// mute - mute sound output
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::mute(bool mute, u8 reason)
|
|
{
|
|
if (mute)
|
|
m_muted |= reason;
|
|
else
|
|
m_muted &= ~reason;
|
|
set_attenuation(m_attenuation);
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// reset - reset all sound chips
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::reset()
|
|
{
|
|
// reset all the sound chips
|
|
for (device_sound_interface &sound : sound_interface_iterator(machine().root_device()))
|
|
sound.device().reset();
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// pause - pause sound output
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::pause()
|
|
{
|
|
mute(true, MUTE_REASON_PAUSE);
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// resume - resume sound output
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::resume()
|
|
{
|
|
mute(false, MUTE_REASON_PAUSE);
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// config_load - read and apply data from the
|
|
// configuration file
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::config_load(config_type cfg_type, util::xml::data_node const *parentnode)
|
|
{
|
|
// we only care about game files
|
|
if (cfg_type != config_type::GAME)
|
|
return;
|
|
|
|
// might not have any data
|
|
if (parentnode == nullptr)
|
|
return;
|
|
|
|
// iterate over channel nodes
|
|
for (util::xml::data_node const *channelnode = parentnode->get_child("channel"); channelnode != nullptr; channelnode = channelnode->get_next_sibling("channel"))
|
|
{
|
|
mixer_input info;
|
|
if (indexed_mixer_input(channelnode->get_attribute_int("index", -1), info))
|
|
{
|
|
float defvol = channelnode->get_attribute_float("defvol", 1.0f);
|
|
float newvol = channelnode->get_attribute_float("newvol", -1000.0f);
|
|
if (newvol != -1000.0f)
|
|
info.stream->set_user_gain(info.inputnum, newvol / defvol);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// config_save - save data to the configuration
|
|
// file
|
|
//-------------------------------------------------
|
|
|
|
void sound_manager::config_save(config_type cfg_type, util::xml::data_node *parentnode)
|
|
{
|
|
// we only care about game files
|
|
if (cfg_type != config_type::GAME)
|
|
return;
|
|
|
|
// iterate over mixer channels
|
|
if (parentnode != nullptr)
|
|
for (int mixernum = 0; ; mixernum++)
|
|
{
|
|
mixer_input info;
|
|
if (!indexed_mixer_input(mixernum, info))
|
|
break;
|
|
float newvol = info.stream->user_gain(info.inputnum);
|
|
|
|
if (newvol != 1.0f)
|
|
{
|
|
util::xml::data_node *const channelnode = parentnode->add_child("channel", nullptr);
|
|
if (channelnode != nullptr)
|
|
{
|
|
channelnode->set_attribute_int("index", mixernum);
|
|
channelnode->set_attribute_float("newvol", newvol);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
//-------------------------------------------------
|
|
// update - mix everything down to its final form
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// and send it to the OSD layer
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//-------------------------------------------------
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void sound_manager::update(void *ptr, int param)
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{
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VPRINTF(("sound_update\n"));
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g_profiler.start(PROFILER_SOUND);
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// force all the speaker streams to generate the proper number of samples
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int samples_this_update = 0;
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for (speaker_device &speaker : speaker_device_iterator(machine().root_device()))
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speaker.mix(&m_leftmix[0], &m_rightmix[0], samples_this_update, (m_muted & MUTE_REASON_SYSTEM));
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// now downmix the final result
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u32 finalmix_step = machine().video().speed_factor();
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u32 finalmix_offset = 0;
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s16 *finalmix = &m_finalmix[0];
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int sample;
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for (sample = m_finalmix_leftover; sample < samples_this_update * 1000; sample += finalmix_step)
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{
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int sampindex = sample / 1000;
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// clamp the left side
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s32 samp = m_leftmix[sampindex];
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if (samp < -32768)
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samp = -32768;
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else if (samp > 32767)
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samp = 32767;
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finalmix[finalmix_offset++] = samp;
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// clamp the right side
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samp = m_rightmix[sampindex];
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if (samp < -32768)
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samp = -32768;
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else if (samp > 32767)
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samp = 32767;
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finalmix[finalmix_offset++] = samp;
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}
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m_finalmix_leftover = sample - samples_this_update * 1000;
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// play the result
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if (finalmix_offset > 0)
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{
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if (!m_nosound_mode)
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machine().osd().update_audio_stream(finalmix, finalmix_offset / 2);
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machine().osd().add_audio_to_recording(finalmix, finalmix_offset / 2);
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machine().video().add_sound_to_recording(finalmix, finalmix_offset / 2);
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if (m_wavfile != nullptr)
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wav_add_data_16(m_wavfile, finalmix, finalmix_offset);
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}
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|
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// see if we ticked over to the next second
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|
attotime curtime = machine().time();
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|
bool second_tick = false;
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|
if (curtime.seconds() != m_last_update.seconds())
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|
{
|
|
assert(curtime.seconds() == m_last_update.seconds() + 1);
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|
second_tick = true;
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|
}
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|
|
|
// iterate over all the streams and update them
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for (auto &stream : m_stream_list)
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|
stream->update_with_accounting(second_tick);
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|
|
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// remember the update time
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|
m_last_update = curtime;
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|
|
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// update sample rates if they have changed
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|
for (auto &stream : m_stream_list)
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stream->apply_sample_rate_changes();
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g_profiler.stop();
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|
}
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